I have several digitized vinyls in FLAC, at 24 bit, 96-192 khz. Is it worth all that sample rate? I was thinking of making CD quality (16/44.1) FLACs or even mp3 320. I doubt I'll ever in my life spend more than, idk, 200 usd in headphones, so the over-quality looks a waste. What do you say? I'm far from an audiophile and I feel pretty happy with mp3 320 (I can't even tell from FLAC).

Anyway, I need to re-encode that at least for my mobile devices. Should I have any special care when converting? I use fre:ac (LAME 3.99.5). Should I just set 44.1 khz when encoding there and done? Or should I downsample with another tool and then encode? I just tried encoding something setting 44.1 in LAME and the result sounds the same as the original 24/192 (with my crappy headphones at least).

The same goes for FLAC or ogg (I tend to use the latter for my DAP). Just setting the sample rate when encoding is enough or is it better to do it with another program? (I doubt, but I wouldn't like regretting later because I made a crappy conversion).

Thanks!

(maybe this should go in the "vinyl" forum, but my concern is more with those high sample rates and downsampling, regardless the source)

I have several digitized vinyls in FLAC, at 24 bit, 96-192 khz. Is it worth all that sample rate?

Certainly not. Compared 96 kHz gives you a frequency range up to 48 kHz and 192 the double. That's 1 resp. 2 octaves above what you can hear (at least!). And LPs don't have 24 bits resolution, so you are carefully preserving the noise floor

The case for lossless is that you don't want to to this ripping job again; should you ever get into issues with your mp3's, then transcoding from lossy to lossy could be audible even at bitrates where you normally don't hear any difference (which are lower than you think!).

Thinking about it I'll stay with FLAC definitely, finding vinyls from 40 years ago (and in good condition) from a foreign country is no easy task. And then decent equipment to make it digital.

But converting to a more "sane" sampling/bit depth, while preserving the quality that could be perceived with good audio equipment, seems a good idea. Any suggestion? Should I go for CD standard, 16/44.1? Or stay at 96 kHz? And for bit depth? All FLACs I have are 192 khz (some 96) with 32 bits floating point. A bit overkill I understand.

Btw, I use foobar with the SoX Resampler plugin for converting. In the output section I can set the bit depth. Should I have dither on if I reduce it? ("always" is the option foobar has).

Sorry for the double post, I can't see an 'edit' button on my post anymore

Anyway, I just checked other FLACs I have that are supposedly 24 bit, but in Audacity show up as 32 floating point. Foobar says 24, as it should be. Forget every time I said "32 bits" before, it's 24 actually

I'm thinking of making 96 khz, 24 bits FLACs, keep those and discard the huge 192 khz monster. It's not going as low as 44.1, and the file size is ok for me. What do you say? One question remains: if I downsample from 192 to 96, and then those 96 to 44.1 (I'll keep the 96 khz one as "master" and the other for mobile playback), will I have any [perceptible] loss than if I went from 192 directly to 44.1? I just tried and the graphs look identical in both cases.

Go 44.1 or 48, as those are the well-supported sample rates. (CDs are 44.1 as you know, but the AC97 soundcards came with 48 kHz only and had to resample the 44.1 (at quality that has been discussed ever since). Don't think it is a problem anymore, but ask someone else -- myself I am using an external DAC that eats pretty much everything).)

16 bits is compatible with everything that supports FLAC, it should be sufficient for the human ear, and should be more than plenty for vinyl rips. But in order to get 16 bits, then your files must actually use the most significant ones. Try running a ReplayGain scan and see what the peaks are like.

QUOTE

No way to tell a difference, even in the graphs

Graphs are not really useful (well they may be for certain purposes under certain circumstances, but not this). Gross visual differences may fail to be audible, and pretty significant audible differences won't be detected visually (at least not without practically useless zooming, inspecting a split second at the time).

Try the FAQ for good info and advice. Also this page, which is one of the first google results for "LP noise floor": Transferring LPs to CDR.

Any frequency content in audio above 20kHz is completely inaudible to human listeners. On your LPs, though you're picking up content up to 40kHz, practically all of the content beyond 16kHz is going to be noise, not signal. (Anything above 20kHz is definitely going to be noise). Keeping 96kHz versions is a waste.

The LP noise floor is rather high- maybe -70dB under very good conditions. 12-bit sampling (RMS noise floor of -72dB) would be sufficient for LP use as long as your levels are right (peak signal above -6dB). (12-bit sampling was used for DV but hasn't seen any other widespread use).

You would most likely find that 32kHz 12-bit would sound just as good in regular listening as anything else.

Any extra detail you capture beyond that is basically going to be useful for editing, not listening. (People try a lot of different things to help clean up LP transfers, and leaving yourself a little room in case you want to try that in the future could be nice.) Extra bit depth is usually more helpful for editing than a higher sampling rate. 16-bit is plenty of extra room. There may be a little bit of real signal on your LPs in the 16-20 kHz range, even if it's not really audible. Downsampling to 48 or even 44.1 kHz retains all that and more, and gives you room for a reasonable transition band on top of that.

I'd really recommend you go with 48kHz 16-bit FLAC.

BTW, even if you had high-fidelity digital originals rather than an LP, here's a paper on why- at least for playback purposes- >16 bit sampling is just a waste of storage while extra-high sampling rates are actually *harmful*.

Oh, BTW: resampling in multiple stages is unlikely to cause any trouble since you've got so much more information than is needed to faithfully reproduce the actual signal, as long as you use a good resampler. Depending on your version of Audacity and your settings, you may get inferior resampling, which could start to make an audible difference in repeated downsampling. SoX and libsamplerate both do a better job. SRC Comparisons is a nice site for getting more info, though you have to keep in mind that some of the distortions that appear obvious on their charts are definitely inaudible.

Great answers, thanks. I just read the links you posted. I see now why go for 16 bits 48 khz. The recordings are quite clear, the little cracks and ticks from the LP are quiet enough, barely audible. The conversion from the vinyl is quite good. No further editing seems necessary, so keeping larger files is pointless as I understand.

Now, what's the best way to reduce the bit depth? Dither or not? Is foobar's option in the conversion for the output bit depth enough? I'd like a simple, but effective, solution that can be made in batchs (like SoX resampler in foobar). I'll try a ReplayGain to check the peak levels when I get home. If they stay above -6 db it's ok to go for 16 bits then? What would you do for getting 24 to 16?

For resampling I'll stick with SoX. It appears good enough from what I read.

... Anyway, I just checked other FLACs I have that are supposedly 24 bit, but in Audacity show up as 32 floating point. Foobar says 24, as it should be. ...

The bit depth shown by Audacity is the default format that it converts input to and uses internally. If you want to keep it in 24 bit, for example you're just editing (cutting / pasting) and not doing any math (EQ, level changing etc), you can change the default using "Preferences --> Quality --> Default Sample Format".

If you're already using SoX, no reason to not use it for bit depth conversion too.

CODE

sox in.wav -b 16 out.wav rate 48k

should do the job quite nicely. It will use triangular pdf dither by default; since the LP noise floor is so much higher than that of 16-bit it may not really matter whether you dither or not, but it won't do any harm. Don't bother with noise-shaped dithers.

... Anyway, I just checked other FLACs I have that are supposedly 24 bit, but in Audacity show up as 32 floating point. Foobar says 24, as it should be. ...

The bit depth shown by Audacity is the default format that it converts input to and uses internally. If you want to keep it in 24 bit, for example you're just editing (cutting / pasting) and not doing any math (EQ, level changing etc), you can change the default using "Preferences --> Quality --> Default Sample Format".

Indeed. Everything, even mp3, it takes at 32 bits. Thanks for the tip/

QUOTE (jensend @ Aug 27 2012, 20:09)

If you're already using SoX, no reason to not use it for bit depth conversion too.

CODE

sox in.wav -b 16 out.wav rate 48k

should do the job quite nicely. It will use triangular pdf dither by default; since the LP noise floor is so much higher than that of 16-bit it may not really matter whether you dither or not, but it won't do any harm. Don't bother with noise-shaped dithers.

I was using the SoX plugin in foobar, but that's limited to re-sampling. I'll use the "stand-alone" SoX with that command.

After checking with Replay Gain in foobar all tracks of the LPs are above ~4.30db (that being the lowest value) and the peak or something (a number that appeared beside that, I don't have foobar here to check) was always around 0.99 and 0.95.

Btw, I wonder why so many ppl (almost everyone it seems) rips their vinyls at 24/96... Maybe keep the 24 bits for editing, but why such a high frequency is so wide-spread? From what I see and you told me it's more than clear it's practically useless.

It’s a result of one of audio’s many compound myths: the idea that vinyl as a medium has superior support for high-frequency content + the idea that there would be any reason to preserve such frequencies even if it did.I refer to useful high-frequency content; it has plenty of worthless noise and distortion in those areas, I’m sure.

You’re right that 24 bits can be useful/advisable for editing, but implicit in that statement is the fact that such a high bit-depth is unnecessary for delivery/listening. Along these lines, you might be interested in this very good article by Monty of Xiph/Vorbis fame: 24/192 Music Downloads are Very Silly Indeed

After checking with Replay Gain in foobar all tracks of the LPs are above ~4.30db (that being the lowest value) and the peak or something (a number that appeared beside that, I don't have foobar here to check) was always around 0.99 and 0.95.

That's very good. It means that going for 16 bits will give you 16 bits.

(I think that people tend to believe that as an analogue source doesn't have a fixed resolution, it has infinite resolution.)

There is some use to that over-20kHz stuff, though. If you have lots, it shows that there is mistracking going on. If you have sudden bursts, that's a "pop". It should be possible to use this constructively, but I haven't worked on/with this yet personally.

I have several digitized vinyls in FLAC, at 24 bit, 96-192 khz. Is it worth all that sample rate? I was thinking of making CD quality (16/44.1) FLACs or even mp3 320. I doubt I'll ever in my life spend more than, idk, 200 usd in headphones, so the over-quality looks a waste.

You are ignoring the weakest link in your system which is not the $200 heaphones, but rather your own personal ear/brain system.

QUOTE

What do you say? I'm far from an audiophile and I feel pretty happy with mp3 320 (I can't even tell from FLAC).

Don't feel strange or diminished in any way. Hearing differences between well-made 320k MP3s and FLAC files (which are indistinguishable in every reasonable way from uncompressed wave files) is very difficult - only done by trained listeners using musical selections designed to be difficult to code as MP3s.

Unless you are doing some unusual things with noise reduction, there is no reason to record vinyl in any uncompressed format beyond 16 bits and 44.1 KHz sampling. Nobody hears the difference between that and anything with bigger numbers attached.

Vinyl itself can usually be recorded transparently with 12 or13 bits and 32-36 KHz sampling. Anything beyond that is headroom. So 16/44 is overkill. It works well for CD burning, so use it!

After checking with Replay Gain in foobar all tracks of the LPs are above ~4.30db (that being the lowest value) and the peak or something (a number that appeared beside that, I don't have foobar here to check) was always around 0.99 and 0.95.

Good. To expand a little on what Porcus said about this meaning you really will get 16 bits: as long as your signal peaks above 0.5 of full scale i.e. above -6dB SPL (sound pressure level), you're making use of the most significant bit and therefore getting the dynamic range and noise floor of a 16-bit recording, while if e.g. your signal was always below 1/16 of full scale i.e. below -24dB SPL, the first four bits of every sample would be zero so you'd only be effectively using 12 bits.

QUOTE

Btw, I wonder why so many ppl (almost everyone it seems) rips their vinyls at 24/96... Maybe keep the 24 bits for editing, but why such a high frequency is so wide-spread? From what I see and you told me it's more than clear it's practically useless.

It's just part of the same group confusion that is common among all sorts of audiophiles (not just those dealing with vinyl). There is, however, a real reason why people started making equipment that could record at higher sampling rates in the first place, and that story may help you understand how the confusion among audiophiles started.

Whenever you run many kinds of filters on a signal, to avoid aliasing, pre-echo, and other problems you need a "transition band" between the highest frequencies you are trying to faithfully preserve and the Nyquist frequency (1/2 your sample rate). When the transition band is narrower, the filter has to be more complicated and wait until it's seen more of the signal before it outputs anything (there's kind of a heisenberg type frequency*time inequality here).

For digital offline processing and editing this doesn't really matter so much any more- computers are powerful enough these days to run very complicated filters (like the sinc filters that make SoX's VHQ resampling work) reasonably fast, so we can use very narrow transition bands. But complicated analog filters can be noisy and/or expensive, and if you're doing effects and mixing real-time in the studio, having to wait to look at more of the signal means you're introducing more delay, which may be unacceptable. So for analog and/or real-time use, a transition band from 20kHz to 24kHz, ~1/4 of an octave, may be too narrow, meaning that you can get better results by recording at sampling rates above 48kHz. If you sample at 96kHz, you have a nice wide ~1 1/4 octave transition band from 20kHz to 48kHz, and can therefore use lots of fast, simple filters without introducing artifacts in any of the audible frequencies.

Because recording at 96kHz could in these circumstances lead to a better end result, misguided audiophiles started thinking that a) they wanted to have the original non-downsampled master and b) more kHz is always better.

The LP noise floor is rather high- maybe -70dB under very good conditions. 12-bit sampling (RMS noise floor of -72dB) would be sufficient for LP use as long as your levels are right (peak signal above -6dB). (12-bit sampling was used for DV but hasn't seen any other widespread use).

The SNR of an MP3 is around 25-30dB. Does this mean that I only need to decode them to 6-bit PCM to capture all the details?

I am curious if anyone has ever done a detailed analysis of an LP's SNR in different frequency bands.

Should I care? What do those samples mean? 19 and 16 samples from the 48000 in a second? (each wav is a full disk, an hour long). Should I do something? (I know, "decrease volume", but I wouldn't like to do that for something insignificant). I'm guessing it's just cracks/ticks or noise from the vinyl that have a spike in the audio.

The LP noise floor is rather high- maybe -70dB under very good conditions. 12-bit sampling (RMS noise floor of -72dB) would be sufficient for LP use as long as your levels are right (peak signal above -6dB). (12-bit sampling was used for DV but hasn't seen any other widespread use).

The SNR of an MP3 is around 25-30dB. Does this mean that I only need to decode them to 6-bit PCM to capture all the details?

I am curious if anyone has ever done a detailed analysis of an LP's SNR in different frequency bands.

This just goes to make a point I've made in talks every since about 1990: "SNR is Mostly Harmless"

If it matters of something, the files with clipping have a track gain between -2.50 and -2.95 db and a peak of 0.999900 (that's what foobar says at least, exactly the same for the 3). It's more or less the same as the others that went without warnings.

If it matters of something, the files with clipping have a track gain between -2.50 and -2.95 db and a peak of 0.999900 (that's what foobar says at least, exactly the same for the 3). It's more or less the same as the others that went without warnings.

The track gain is the gain adjustment required to reach the perceptual loudness level recommended in the ReplayGain standard, so it's saying your recordings are 2.5-3 dB too loud according to that standard. For resampling purposes it's usually good to have a maximum level of .95 or lower (close to 0.5dB of headroom).

QUOTE (Taishou @ Aug 28 2012, 19:48)

Should I care? What do those samples mean? 19 and 16 samples from the 48000 in a second? (each wav is a full disk, an hour long). Should I do something? (I know, "decrease volume", but I wouldn't like to do that for something insignificant). I'm guessing it's just cracks/ticks or noise from the vinyl that have a spike in the audio.

It's true that clipping a handful of samples in an hour is probably not worth much worrying about, especially if those are already pops/clicks- which aren't really going to be made worse by a little further distortion- rather than real signal. You'll have to decide whether you want to do anything about it or not.

But before you make that decision you should know there's an easy way to do it and do it well. You don't have to choose between just reducing every file's gain by the same amount (too much for some and/or not enough for others) and manually choosing an appropriate gain for each file (very time consuming). Adding -G to your SoX command line i.e. "sox -G ..." will automatically apply just the right amount of negative gain to prevent clipping and won't lose any precision in intermediate steps; see its manual for details.